/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
H A D | jitter_buffer_unittest.cc | 19 #include "webrtc/modules/video_coding/main/source/packet.h" 144 VCMPacket packet; local 145 packet.dataPtr = data_buffer_; 146 bool packet_available = stream_generator_->PopPacket(&packet, index); 151 return jitter_buffer_->InsertPacket(packet, &retransmitted); 155 VCMPacket packet; local 156 packet.dataPtr = data_buffer_; 157 bool packet_available = stream_generator_->GetPacket(&packet, index); 162 return jitter_buffer_->InsertPacket(packet, &retransmitted); 268 // Insert the packet t 1863 VCMPacket packet; local 1899 VCMPacket packet; local 1905 VCMPacket packet; local [all...] |
H A D | receiver.cc | 74 int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, argument 77 // Insert the packet into the jitter buffer. The packet can either be empty or 80 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet, 93 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds()); 267 // dual decoder has caught up with the decoder decoding with packet losses.
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H A D | receiver_unittest.cc | 15 #include "webrtc/modules/video_coding/main/source/packet.h" 45 VCMPacket packet; local 46 packet.dataPtr = data_buffer_; 47 bool packet_available = stream_generator_->GetPacket(&packet, index); 52 return receiver_.InsertPacket(packet, 640, 480); 56 VCMPacket packet; local 57 packet.dataPtr = data_buffer_; 58 bool packet_available = stream_generator_->PopPacket(&packet, index); 62 return receiver_.InsertPacket(packet, kWidth, kHeight); 74 // Drop the second packet 75 VCMPacket packet; local [all...] |
H A D | session_info.cc | 13 #include "webrtc/modules/video_coding/main/source/packet.h" 126 VCMPacket& packet = *packet_it; local 129 // Calculate the offset into the frame buffer for this packet. 134 // Set the data pointer to pointing to the start of this packet in the 136 const uint8_t* packet_buffer = packet.dataPtr; 137 packet.dataPtr = frame_buffer + offset; 143 if (packet.codecSpecificHeader.codec == kRtpVideoH264 && 144 packet.codecSpecificHeader.codecHeader.H264.stap_a) { 147 while (nalu_ptr < packet_buffer + packet.sizeBytes) { 150 length + (packet 438 InsertPacket(const VCMPacket& packet, uint8_t* frame_buffer, VCMDecodeErrorMode decode_error_mode, const FrameData& frame_data) argument [all...] |
H A D | session_info_unittest.cc | 15 #include "webrtc/modules/video_coding/main/source/packet.h" 176 // Insert empty packet which will be the new high sequence number. 349 // Insert an older packet with a first packet set. 427 // Insert out of bound regular packets, and then the first and last packet. 438 // Insert an older packet with a first packet set. 494 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local 496 EXPECT_EQ(session_.InsertPacket(*packet, 501 delete packet; 551 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local 622 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local 693 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local 765 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local 836 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local 936 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/test/ |
H A D | stream_generator.cc | 18 #include "webrtc/modules/video_coding/main/source/packet.h" 69 VCMPacket packet; local 70 packet.seqNum = sequence_number; 71 packet.timestamp = timestamp; 72 packet.frameType = type; 73 packet.isFirstPacket = first_packet; 74 packet.markerBit = marker_bit; 75 packet.sizeBytes = size; 76 packet.dataPtr = packet_buffer; 77 if (packet 86 PopPacket(VCMPacket* packet, int index) argument 96 GetPacket(VCMPacket* packet, int index) argument 105 NextPacket(VCMPacket* packet) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | mt_test_common.cc | 55 // Insert outgoing packet into list 61 // Simulate receive time = network delay + packet jitter 77 RtpPacket* packet = NULL; local 83 // Take first packet in list 84 packet = _rtpPackets.front(); 85 int64_t timeToReceive = packet->receiveTime - now; 96 if (!parser->Parse(packet->data, packet->length, &header)) { 97 delete packet; 105 if (!rtp_receiver_->IncomingRtpPacket(header, packet [all...] |
H A D | rtp_player.cc | 95 void AddPacket(RawRtpPacket* packet) { argument 96 assert(packet); 97 printf("Throw: %08x:%u\n", packet->ssrc(), packet->seq_num()); 100 fprintf(debug_file_, "%u Lost packet: %u\n", loss_count_, 101 packet->seq_num()); 103 packets_.push_back(packet); 112 RawRtpPacket* packet = *it; local 113 if (ssrc == packet->ssrc() && resendSeqNum == packet 129 RawRtpPacket* packet = *it; local 151 LogPacketResent(RawRtpPacket* packet) argument [all...] |
H A D | test_callbacks.cc | 64 // will call the VCMReceiver input packet 241 // Take first packet in list 265 // Insert outgoing packet into list 271 // Simulate receive time = network delay + packet jitter 282 RtpPacket* packet = NULL; local 286 // Take first packet in list 287 packet = _rtpPackets.front(); 288 int64_t timeToReceive = packet->receiveTime - now; 300 if (!parser->Parse(packet->data, packet [all...] |
/external/chromium_org/third_party/webrtc/test/ |
H A D | fake_network_pipe.cc | 65 // The packet data. 69 // The time the packet was sent out on the network. 71 // The time the packet should arrive at the reciver. 114 // Too many packet on the link, drop this one. 133 NetworkPacket* packet = new NetworkPacket(data, data_length, time_now, local 135 capacity_link_.push(packet); 163 // Time to get this packet. 164 NetworkPacket* packet = capacity_link_.front(); local 169 delete packet; 174 // earlier than the last packet i 193 NetworkPacket* packet = delay_link_.front(); local 204 NetworkPacket* packet = packets_to_deliver.front(); local [all...] |
H A D | null_transport.cc | 15 bool NullTransport::SendRtp(const uint8_t* packet, size_t length) { argument 19 bool NullTransport::SendRtcp(const uint8_t* packet, size_t length) { argument
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H A D | rtcp_packet_parser.cc | 21 const uint8_t* packet = static_cast<const uint8_t*>(data); local 22 RTCPUtility::RTCPParserV2 parser(packet, len, true);
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H A D | rtp_file_reader_unittest.cc | 32 test::RtpFileReader::Packet packet; local 34 while (rtp_packet_source_->NextPacket(&packet)) 62 test::RtpFileReader::Packet packet; local 63 while (rtp_packet_source_->NextPacket(&packet)) 70 test::RtpFileReader::Packet packet; local 71 while (rtp_packet_source_->NextPacket(&packet)) { 72 RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
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/external/chromium_org/third_party/webrtc/video/ |
H A D | call.cc | 104 virtual DeliveryStatus DeliverPacket(const uint8_t* packet, 110 DeliveryStatus DeliverRtcp(const uint8_t* packet, size_t length); 111 DeliveryStatus DeliverRtp(const uint8_t* packet, size_t length); 346 PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet, argument 349 // Do NOT broadcast! Also make sure it's a valid packet. 351 // there's no receiver of the packet. 359 if (it->second->DeliverRtcp(packet, length)) 370 if (it->second->DeliverRtcp(packet, length)) 377 PacketReceiver::DeliveryStatus Call::DeliverRtp(const uint8_t* packet, argument 383 const uint8_t* ptr = &packet[ 397 DeliverPacket(const uint8_t* packet, size_t length) argument [all...] |
H A D | call_perf_tests.cc | 64 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE { 65 RTCPUtility::RTCPParserV2 parser(packet, length, true); 72 const RTCPUtility::RTCPPacket& packet = parser.Packet(); variable 74 packet.SR.NTPMostSignificant, 75 packet.SR.NTPLeastSignificant, 76 packet.SR.RTPTimestamp); 197 virtual DeliveryStatus DeliverPacket(const uint8_t* packet, 200 if (parser_->IsRtcp(packet, static_cast<int>(length))) { 202 channel_, packet, static_cast<unsigned int>(length)); 205 channel_, packet, static_cas 374 OnSendRtp(const uint8_t* packet, size_t length) argument [all...] |
H A D | end_to_end_tests.cc | 58 virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { 63 virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE { 281 virtual Action OnReceiveRtcp(const uint8_t* packet, 283 RTCPUtility::RTCPParserV2 parser(packet, length, true); 286 ssrc |= static_cast<uint32_t>(packet[4]) << 24; 287 ssrc |= static_cast<uint32_t>(packet[5]) << 16; 288 ssrc |= static_cast<uint32_t>(packet[6]) << 8; 289 ssrc |= static_cast<uint32_t>(packet[7]) << 0; 298 << "Timed out while waiting for a receiver RTCP packet to be sent."; 319 virtual Action OnSendRtp(const uint8_t* packet, size_ 1125 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local 1129 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local 1185 OnSendRtcp(const uint8_t* packet, size_t length) argument [all...] |
H A D | rampup_tests.cc | 106 // Just trigger if there was any rtx padding packet. 116 bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) { argument 119 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); 140 packet, 153 bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) { argument 265 const uint8_t* packet, size_t length) { 268 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); 279 bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) { argument 264 DeliverPacket( const uint8_t* packet, size_t length) argument
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H A D | replay.cc | 239 test::RtpFileReader::Packet packet; local 240 if (!rtp_reader->NextPacket(&packet)) 243 switch (call->Receiver()->DeliverPacket(packet.data, packet.length)) { 249 parser->Parse(packet.data, packet.length, &header); 259 if (last_time_ms != 0 && last_time_ms != packet.time_ms) { 260 SleepMs(packet.time_ms - last_time_ms); 262 last_time_ms = packet.time_ms;
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H A D | transport_adapter.cc | 20 const void* packet, 25 bool success = transport_->SendRtp(static_cast<const uint8_t*>(packet), 31 const void* packet, 36 bool success = transport_->SendRtcp(static_cast<const uint8_t*>(packet), 19 SendPacket(int , const void* packet, int length) argument 30 SendRTCPPacket(int , const void* packet, int length) argument
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H A D | video_receive_stream.cc | 232 bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { argument 234 channel_, packet, static_cast<int>(length)) == 0; 237 bool VideoReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) { argument 239 channel_, packet, static_cast<int>(length), PacketTime()) == 0;
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H A D | video_send_stream.cc | 196 // 28 to match packet overhead in ModuleRtpRtcpImpl. 404 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { argument 406 channel_, packet, static_cast<int>(length)) == 0;
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/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
H A D | vie_autotest_network.cc | 218 // Create a empty RTP packet. 219 unsigned char packet[3000]; local 220 memset(packet, 0, sizeof(packet)); 221 packet[0] = 0x80; // V=2, P=0, X=0, CC=0 222 packet[1] = 0x7C; // M=0, PT = 124 (I420) 224 // Create a empty RTCP app packet. 233 tbChannel.videoChannel, packet, 1500)); 238 tbChannel.videoChannel, packet, 1500)); 242 tbChannel.videoChannel, packet, 1 [all...] |
/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/testbed/ |
H A D | tb_external_transport.cc | 383 VideoPacket* packet = NULL; local 388 // Take first packet in queue 389 packet = _rtpPackets.front(); 391 if (packet) 393 timeToReceive = packet->receiveTime - NowMs(); 413 if (packet) 418 ssrc = ((packet->packetBuffer[8]) << 24); 419 ssrc += (packet->packetBuffer[9] << 16); 420 ssrc += (packet->packetBuffer[10] << 8); 421 ssrc += packet [all...] |
/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_receiver.cc | 202 // Only forward if the incoming packet *and* the channel are both configured 248 // that the first packet is included in the stats). 254 bool ViEReceiver::ReceivePacket(const uint8_t* packet, argument 259 return ParseAndHandleEncapsulatingHeader(packet, packet_length, header); 261 const uint8_t* payload = packet + header.headerLength; 273 bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet, argument 278 if (packet[header.headerLength] == ulpfec_pt) 281 header, packet, packet_length, ulpfec_pt) != 0) { 287 // This is an empty packet and should be silently dropped before trying to 298 LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | channel.cc | 198 "Channel::SendPacket() failed to send RTP packet due to" 206 // Dump the RTP packet to a file (if RTP dump is enabled). 242 "Channel::SendRTCPPacket() failed to send RTCP packet" 250 // Dump the RTCP packet to a file (if RTP dump is enabled). 482 // It is possible that the connection is alive even if no RTP packet has 484 // and a low SID-packet update rate. 523 // packet as discarded. 526 "received packet is discarded since playing is not" 543 // Update the packet delay. 1471 "SetSendCodec() failed to set audio packet siz 1804 ReceivePacket(const uint8_t* packet, int packet_length, const RTPHeader& header, bool in_order) argument 1823 HandleEncapsulation(const uint8_t* packet, int packet_length, const RTPHeader& header) argument [all...] |